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QUIC: A UDP-Based Multiplexed and Secure TransportGooglejri@google.comMozillamartin.thomson@gmail.comTransport
QUICThis document defines the core of the QUIC transport protocol. This document
describes connection establishment, packet format, multiplexing and reliability.
Accompanying documents describe the cryptographic handshake and loss detection.Discussion of this draft takes place on the QUIC working group mailing list
(quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic.Working Group information can be found at https://github.com/quicwg; source
code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/transport.QUIC is a multiplexed and secure transport protocol that runs on top of UDP.
QUIC aims to provide a flexible set of features that allow it to be a
general-purpose transport for multiple applications.QUIC implements techniques learned from experience with TCP, SCTP and other
transport protocols. Using UDP as the substrate, QUIC seeks to be compatible
with legacy clients and middleboxes. QUIC authenticates all of its headers and
encrypts most of the data it exchanges, including its signaling. This allows
the protocol to evolve without incurring a dependency on upgrades to
middleboxes.This document describes the core QUIC protocol, including the conceptual design,
wire format, and mechanisms of the QUIC protocol for connection establishment,
stream multiplexing, stream and connection-level flow control, and data
reliability.Accompanying documents describe QUIC’s loss detection and congestion control
, and the use of TLS 1.3 for key negotiation .The words “MUST”, “MUST NOT”, “SHOULD”, and “MAY” are used in this document.
It’s not shouting; when they are capitalized, they have the special meaning
defined in .Definitions of terms that are used in this document:
The endpoint initiating a QUIC connection.
The endpoint accepting incoming QUIC connections.
The client or server end of a connection.
A logical, bi-directional channel of ordered bytes within a QUIC connection.
A conversation between two QUIC endpoints with a single encryption context
that multiplexes streams within it.
The identifier for a QUIC connection.
A well-formed UDP payload that can be parsed by a QUIC receiver. QUIC packet
size in this document refers to the UDP payload size.Packet and frame diagrams use the format described in Section 3.1,
with the following additional conventions:
Indicates that x is optional
Indicates that x is encrypted
Indicates that x is A bits long
Indicates that x is one of A, B, or C bits long
Indicates that x is variable-lengthThis section briefly describes QUIC’s key mechanisms and benefits. Key
strengths of QUIC include:Low-latency connection establishmentMultiplexing without head-of-line blockingAuthenticated and encrypted header and payloadRich signaling for congestion control and loss recoveryStream and connection flow controlConnection migration and resilience to NAT rebindingVersion negotiationQUIC relies on a combined cryptographic and transport handshake for
setting up a secure transport connection. QUIC connections are
expected to commonly use 0-RTT handshakes, meaning that for most QUIC
connections, data can be sent immediately following the client
handshake packet, without waiting for a reply from the server. QUIC
provides a dedicated stream (Stream ID 1) to be used for performing
the cryptographic handshake and QUIC options negotiation. The format
of the QUIC options and parameters used during negotiation are
described in this document, but the handshake protocol that runs on
Stream ID 1 is described in the accompanying cryptographic handshake
draft .When application messages are transported over TCP, independent application
messages can suffer from head-of-line blocking. When an application multiplexes
many streams atop TCP’s single-bytestream abstraction, a loss of a TCP segment
results in blocking of all subsequent segments until a retransmission arrives,
irrespective of the application streams that are encapsulated in subsequent
segments. QUIC ensures that lost packets carrying data for an individual stream
only impact that specific stream. Data received on other streams can continue
to be reassembled and delivered to the application.QUIC’s packet framing and acknowledgments carry rich information that help both
congestion control and loss recovery in fundamental ways. Each QUIC packet
carries a new packet number, including those carrying retransmitted data. This
obviates the need for a separate mechanism to distinguish acknowledgments for
retransmissions from those for original transmissions, avoiding TCP’s
retransmission ambiguity problem. QUIC acknowledgments also explicitly encode
the delay between the receipt of a packet and its acknowledgment being sent, and
together with the monotonically-increasing packet numbers, this allows for
precise network roundtrip-time (RTT) calculation. QUIC’s ACK frames support up
to 256 ACK blocks, so QUIC is more resilient to reordering than TCP with SACK
support, as well as able to keep more bytes on the wire when there is reordering
or loss.QUIC implements stream- and connection-level flow control, closely following
HTTP/2’s flow control mechanisms. At a high level, a QUIC receiver advertises
the absolute byte offset within each stream up to which the receiver is willing
to receive data. As data is sent, received, and delivered on a particular
stream, the receiver sends WINDOW_UPDATE frames that increase the advertised
offset limit for that stream, allowing the peer to send more data on that
stream. In addition to this stream-level flow control, QUIC implements
connection-level flow control to limit the aggregate buffer that a QUIC receiver
is willing to allocate to all streams on a connection. Connection-level flow
control works in the same way as stream-level flow control, but the bytes
delivered and highest received offset are all aggregates across all streams.TCP headers appear in plaintext on the wire and are not authenticated, causing a
plethora of injection and header manipulation issues for TCP, such as
receive-window manipulation and sequence-number overwriting. While some of
these are mechanisms used by middleboxes to improve TCP performance, others are
active attacks. Even “performance-enhancing” middleboxes that routinely
interpose on the transport state machine end up limiting the evolvability of the
transport protocol, as has been observed in the design of MPTCP and
in its subsequent deployability issues.Generally, QUIC packets are always authenticated and the payload is typically
fully encrypted. The parts of the packet header which are not encrypted are
still authenticated by the receiver, so as to thwart any packet injection or
manipulation by third parties. Some early handshake packets, such as the
Version Negotiation packet, are not encrypted, but information sent in these
unencrypted handshake packets is later verified as part of cryptographic
processing.PUBLIC_RESET packets that reset a connection are currently not authenticated.QUIC connections are identified by a 64-bit Connection ID, randomly generated by
the client. QUIC’s consistent connection ID allows connections to survive
changes to the client’s IP and port, such as those caused by NAT rebindings or
by the client changing network connectivity to a new address. QUIC provides
automatic cryptographic verification of a rebound client, since the client
continues to use the same session key for encrypting and decrypting packets.
The consistent connection ID can be used to allow migration of the connection to
a new server IP address as well, since the Connection ID remains consistent
across changes in the client’s and the server’s network addresses.QUIC version negotiation allows for multiple versions of the protocol to be
deployed and used concurrently. Version negotiation is described in
.QUIC versions are identified using a 32-bit value.The version 0x00000000 is reserved to represent an invalid version. This
version of the specification is identified by the number 0x00000001.Versions with the most significant 16 bits of the version number cleared are
reserved for use in future IETF consensus documents.Versions that follow the pattern 0x?a?a?a?a are reserved for use in forcing
version negotiation to be exercised. That is, any version number where the low
four bits of all octets is 1010 (in binary). A client or server MAY advertise
support for any of these reserved versions.Reserved version numbers will probably never represent a real protocol; a client
MAY use one of these version numbers with the expectation that the server will
initiate version negotiation; a server MAY advertise support for one of these
versions and can expect that clients ignore the value.[[RFC editor: please remove the remainder of this section before
publication.]]The version number for the final version of this specification (0x00000001), is
reserved for the version of the protocol that is published as an RFC.Version numbers used to identify IETF drafts are created by adding the draft
number to 0xff000000. For example, draft-ietf-quic-transport-13 would be
identified as 0xff00000D.Implementors are encouraged to register version numbers of QUIC that they
are using for private experimentation on the
github wiki.We first describe QUIC’s packet types and their formats, since some are
referenced in subsequent mechanisms.All numeric values are encoded in network byte order (that is, big-endian) and
all field sizes are in bits. When discussing individual bits of fields, the
least significant bit is referred to as bit 0. Hexadecimal notation is used for
describing the value of fields.Any QUIC packet has either a long or a short header, as indicated by the Header
Form bit. Long headers are expected to be used early in the connection before
version negotiation and establishment of 1-RTT keys, and for public resets.
Short headers are minimal version-specific headers, which can be used after
version negotiation and 1-RTT keys are established.Long headers are used for packets that are sent prior to the completion of
version negotiation and establishment of 1-RTT keys. Once both conditions are
met, a sender SHOULD switch to sending short-form headers. While inefficient,
long headers MAY be used for packets encrypted with 1-RTT keys. The long form
allows for special packets, such as the Version Negotiation and the Public Reset
packets to be represented in this uniform fixed-length packet format. A long
header contains the following fields:
The most significant bit (0x80) of the first octet is set to 1 for long
headers and 0 for short headers.
The remaining seven bits of first octet of a long packet is the packet type.
This field can indicate one of 128 packet types. The types specified for this
version are listed in .
Octets 1 through 8 contain the connection ID. describes the
use of this field in more detail.
Octets 9 to 12 contain the packet number. {{packet-numbers} describes the use
of packet numbers.
Octets 13 to 16 contain the selected protocol version. This field indicates
which version of QUIC is in use and determines how the rest of the protocol
fields are interpreted.
Octets from 17 onwards (the rest of QUIC packet) are the payload of the
packet.The following packet types are defined:TypeNameSection01Version Negotiation02Client Cleartext03Non-Final Server Cleartext04Final Server Cleartext050-RTT Encrypted061-RTT Encrypted (key phase 0)071-RTT Encrypted (key phase 1)08Public ResetThe header form, packet type, connection ID, packet number and version fields of
a long header packet are version-independent. The types of packets defined in
are version-specific. See for
details on how packets from different versions of QUIC are interpreted.(TODO: Should the list of packet types be version-independent?)The interpretation of the fields and the payload are specific to a version and
packet type. Type-specific semantics for this version are described in
, , , and
.The short header can be used after the version and 1-RTT keys are negotiated.
This header form has the following fields:
The most significant bit (0x80) of the first octet of a packet is the header
form. This bit is set to 0 for the short header.
The second bit (0x40) of the first octet indicates whether the Connection ID
field is present. If set to 1, then the Connection ID field is present; if
set to 0, the Connection ID field is omitted.
The third bit (0x20) of the first octet indicates the key phase, which allows
a recipient of a packet to identify the packet protection keys that are used
to protect the packet. See for details.
The remaining 5 bits of the first octet include one of 32 packet types.
lists the types that are defined for short packets.
If the Connection ID Flag is set, a connection ID occupies octets 1 through 8
of the packet. See for more details.
The length of the packet number field depends on the packet type. This field
can be 1, 2 or 4 octets long depending on the short packet type.
Packets with a short header always include a 1-RTT protected payload.The packet type in a short header currently determines only the size of the
packet number field. Additional types can be used to signal the presence of
other fields.TypePacket Number Size011 octet022 octets034 octetsThe header form, connection ID flag and connection ID of a short header packet
are version-independent. The remaining fields are specific to the selected QUIC
version. See for details on how packets from different
versions of QUIC are interpreted.A Version Negotiation packet is sent only by servers and is a response to a
client packet of an unsupported version. It uses a long header and contains:Octet 0: 0x81Octets 1-8: Connection ID (echoed)Octets 9-12: Packet Number (echoed)Octets 13-16: Version (echoed)Octets 17+: PayloadThe payload of the Version Negotiation packet is a list of 32-bit versions which
the server supports, as shown below.See for a description of the version negotiation
process.Cleartext packets are sent during the handshake prior to key negotiation. A
Client Cleartext packet contains:Octet 0: 0x82Octets 1-8: Connection ID (initial)Octets 9-12: Packet numberOctets 13-16: VersionOctets 17+: PayloadNon-Final Server Cleartext packets contain:Octet 0: 0x83Octets 1-8: Connection ID (echoed)Octets 9-12: Packet NumberOctets 13-16: VersionOctets 17+: PayloadFinal Server Cleartext packets contains:Octet 0: 0x84Octets 1-8: Connection ID (final)Octets 9-12: Packet NumberOctets 13-16: VersionOctets 17+: PayloadThe client MUST choose a random 64-bit value and use it as the initial
Connection ID in all packets until the server replies with the final Connection
ID. The server echoes the client’s Connection ID in Non-Final Server Cleartext
packets. The first Final Server Cleartext and all subsequent packets MUST use
the final Connection ID, as described in .The payload of a Cleartext packet consists of a sequence of frames, as described
in .(TODO: Add hash before frames.)Packets encrypted with either 0-RTT or 1-RTT keys may be sent with long headers.
Different packet types explicitly indicate the encryption level for ease of
decryption. These packets contain:Octet 0: 0x85, 0x86 or 0x87Octets 1-8: Connection ID (initial or final)Octets 9-12: Packet NumberOctets 13-16: VersionOctets 17+: Encrypted PayloadA first octet of 0x85 indicates a 0-RTT packet. After the 1-RTT keys are
established, key phases are used by the QUIC packet protection to identify the
correct packet protection keys. The initial key phase is 0. See for
more details.The encrypted payload is both authenticated and encrypted using packet
protection keys. describes packet protection in detail. After
decryption, the plaintext consists of a sequence of frames, as described in
.A Public Reset packet is only sent by servers and is used to abruptly terminate
communications. Public Reset is provided as an option of last resort for a
server that does not have access to the state of a connection. This is intended
for use by a server that has lost state (for example, through a crash or
outage). A server that wishes to communicate a fatal connection error MUST use a
CONNECTION_CLOSE frame if it has sufficient state to do so.A Public Reset packet contains:Octet 0: 0x88Octets 1-8: Echoed data (octets 1-8 of received packet)Octets 9-12: Echoed data (octets 9-12 of received packet)Octets 13-16: VersionOctets 17+: Public Reset ProofFor a client that sends a connection ID on every packet, the Connection ID field
is simply an echo of the initial Connection ID, and the Packet Number field
includes an echo of the client’s packet number (and, depending on the client’s
packet number length, 0, 2, or 3 additional octets from the client’s packet).A Public Reset packet sent by a server indicates that it does not have the
state necessary to continue with a connection. In this case, the server will
include the fields that prove that it originally participated in the connection
(see for details).Upon receipt of a Public Reset packet that contains a valid proof, a client MUST
tear down state associated with the connection. The client MUST then cease
sending packets on the connection and SHOULD discard any subsequent packets that
arrive. A Public Reset that does not contain a valid proof MUST be ignored.TODO: Details to be added.QUIC connections are identified by their 64-bit Connection ID. All long headers
contain a Connection ID. Short headers indicate the presence of a Connection ID
using the CONNECTION_ID flag. When present, the Connection ID is in the same
location in all packet headers, making it straightforward for middleboxes, such
as load balancers, to locate and use it.When a connection is initiated, the client MUST choose a random value and use it
as the initial Connection ID until the final value is available. The initial
Connection ID is a suggestion to the server. The server echoes this value in all
packets until the handshake is successful (see ). On a successful
handshake, the server MUST select the final Connection ID for the connection and
use it in Final Server Cleartext packets. This final Connection ID MAY be the
one proposed by the client or MAY be a new server-selected value. All subsequent
packets from the server MUST contain this value. On handshake completion, the
client MUST switch to using the final Connection ID for all subsequent
packets.Thus, all Client Cleartext packets, 0-RTT Encrypted packets, and Non-Final
Server Cleartext packets MUST use the client’s randomly-generated initial
Connection ID. Final Server Cleartext packets, 1-RTT Encrypted packets, and all
short-header packets MUST use the final Connection ID.The packet number is a 64-bit unsigned number and is used as part of a
cryptographic nonce for packet encryption. Each endpoint maintains a separate
packet number for sending and receiving. The packet number for sending MUST
increase by at least one after sending any packet.A QUIC endpoint MUST NOT reuse a packet number within the same connection (that
is, under the same cryptographic keys). If the packet number for sending
reaches 2^64 - 1, the sender MUST close the connection by sending a
CONNECTION_CLOSE frame with the error code QUIC_SEQUENCE_NUMBER_LIMIT_REACHED
(connection termination is described in .)To reduce the number of bits required to represent the packet number over the
wire, only the least significant bits of the packet number are transmitted over
the wire, up to 32 bits. The actual packet number for each packet is
reconstructed at the receiver based on the largest packet number received on a
successfully authenticated packet.A packet number is decoded by finding the packet number value that is closest to
the next expected packet. The next expected packet is the highest received
packet number plus one. For example, if the highest successfully authenticated
packet had a packet number of 0xaa82f30e, then a packet containing a 16-bit
value of 0x1f94 will be decoded as 0xaa831f94.The sender MUST use a packet number size able to represent more than twice as
large a range than the difference between the largest acknowledged packet and
packet number being sent. A peer receiving the packet will then correctly
decode the packet number, unless the packet is delayed in transit such that it
arrives after many higher-numbered packets have been received. An endpoint MAY
use a larger packet number size to safeguard against such reordering.As a result, the size of the packet number encoding is at least one more than
the base 2 logarithm of the number of contiguous unacknowledged packet numbers,
including the new packet.For example, if an endpoint has received an acknowledgment for packet 0x6afa2f,
sending a packet with a number of 0x6b4264 requires a 16-bit or larger packet
number encoding; whereas a 32-bit packet number is needed to send a packet with
a number of 0x6bc107.The initial value for packet number MUST be a 31-bit random number. That is,
the value is selected from an uniform random distribution between 0 and 2^31-1.
provides guidance on the generation of random values.The first set of packets sent by an endpoint MUST include the low 32-bits of the
packet number. Once any packet has been acknowledged, subsequent packets can
use a shorter packet number encoding.Between different versions the following things are guaranteed to remain
constant:the location of the header form flag,the location of the Connection ID flag in short headers,the location and size of the Connection ID field in both header forms,the location and size of the Version field in long headers, andthe location and size of the Packet Number field in long headers.Implementations MUST assume that an unsupported version uses an unknown packet
format. All other fields MUST be ignored when processing a packet that contains
an unsupported version.The payload of cleartext packets and the plaintext after decryption of encrypted
payloads consists of a sequence of frames, as shown in .Encrypted payloads MUST contain at least one frame, and MAY contain multiple
frames and multiple frame types.Frames MUST fit within a single QUIC packet and MUST NOT span a QUIC packet
boundary. Each frame begins with a Frame Type byte, indicating its type,
followed by additional type-dependent fields:Frame types are listed in . Note that the Frame Type byte in
STREAM and ACK frames is used to carry other frame-specific flags. For all
other frames, the Frame Type byte simply identifies the frame. These frames are
explained in more detail as they are referenced later in the document.Type-field valueFrame typeDefinition0x00PADDING0x01RST_STREAM0x02CONNECTION_CLOSE0x03GOAWAY0x04WINDOW_UPDATE0x05BLOCKED0x07PING0x40 - 0x7fACK0x80 - 0xffSTREAMA QUIC connection is a single conversation between two QUIC endpoints. QUIC’s
connection establishment intertwines version negotiation with the cryptographic
and transport handshakes to reduce connection establishment latency, as
described in . Once established, a connection may migrate to a
different IP or port at either endpoint, due to NAT rebinding or mobility, as
described in . Finally a connection may be terminated by either
endpoint, as described in .QUIC’s connection establishment begins with version negotiation, since all
communication between the endpoints, including packet and frame formats, relies
on the two endpoints agreeing on a version.A QUIC connection begins with a client sending a handshake packet. The details
of the handshake mechanisms are described in , but all of the
initial packets sent from the client to the server MUST use the long header
format and MUST specify the version of the protocol being used.When the server receives a packet from a client with the long header format, it
compares the client’s version to the versions it supports.If the version selected by the client is not acceptable to the server, the
server discards the incoming packet and responds with a Version Negotiation
packet (). This includes a list of versions that the server
will accept. A server MUST send a Version Negotiation packet for every packet
that it receives with an unacceptable version.If the packet contains a version that is acceptable to the server, the server
proceeds with the handshake (). This commits the server to the
version that the client selected.When the client receives a Version Negotiation packet from the server, it should
select an acceptable protocol version. If the server lists an acceptable
version, the client selects that version and reattempts to create a connection
using that version. Though the contents of a packet might not change in
response to version negotiation, a client MUST increase the packet number it
uses on every packet it sends. Packets MUST continue to use long headers and
MUST include the new negotiated protocol version.The client MUST use the long header format and include its selected version on
all packets until it has 1-RTT keys and it has received a packet from the server
which is not a Version Negotiation packet.A client MUST NOT change the version it uses unless it is in response to a
Version Negotiation packet from the server. Once a client receives a packet
from the server which is not a Version Negotiation packet, it MUST ignore
Version Negotiation packets on the same connection.Version negotiation uses unprotected data. The result of the negotiation MUST be
revalidated as part of the cryptographic handshake (see ).For a server to use a new version in the future, clients must correctly handle
unsupported versions. To help ensure this, a server SHOULD include a reserved
version (see ) while generating a Version Negotiation packet.The design of version negotiation permits a server to avoid maintaining state
for packets that it rejects in this fashion. However, when the server generates
a Version Negotiation packet, it cannot randomly generate a reserved version
number. This is because the server is required to include the same value in its
transport parameters (see ). To avoid the selected
version number changing during connection establishment, the reserved version
SHOULD be generated as a function of values that will be available to the server
when later generating its handshake packets.A pseudorandom function that takes client address information (IP and port) and
the client selected version as input would ensure that there is sufficient
variability in the values that a server uses.A client MAY send a packet using a reserved version number. This can be used to
solicit a list of supported versions from a server.QUIC relies on a combined cryptographic and transport handshake to minimize
connection establishment latency. QUIC allocates stream 1 for the cryptographic
handshake. This version of QUIC uses TLS 1.3 .QUIC provides this stream with reliable, ordered delivery of data. In return,
the cryptographic handshake provides QUIC with:authenticated key exchange, where a server is always authenticated,a client is optionally authenticated,every connection produces distinct and unrelated keys,keying material is usable for packet protection for both 0-RTT and 1-RTT
packets, and1-RTT keys have forward secrecyauthenticated values for the transport parameters of the peer (see
)authenticated confirmation of version negotiation (see )authenticated negotiation of an application protocol (TLS uses ALPN
for this purpose)for the server, the ability to carry data that provides assurance that the
client can receive packets that are addressed with the transport address that
is claimed by the client (see )The initial cryptographic handshake message MUST be sent in a single packet.
Any second attempt that is triggered by address validation MUST also be sent
within a single packet. This avoids having to reassemble a message from
multiple packets. Reassembling messages requires that a server maintain state
prior to establishing a connection, exposing the server to a denial of service
risk.The first client packet of the cryptographic handshake protocol MUST fit within
a 1280 octet QUIC packet. This includes overheads that reduce the space
available to the cryptographic handshake protocol.Details of how TLS is integrated with QUIC is provided in more detail in
.During connection establishment, both endpoints make authenticated declarations
of their transport parameters. These declarations are made unilaterally by each
endpoint. Endpoints are required to comply with the restrictions implied by
these parameters; the description of each parameter includes rules for its
handling.The format of the transport parameters is the TransportParameters struct from
. This is described using the presentation
language from Section 3 of .;
} TransportParameter;
struct {
select (Handshake.msg_type) {
case client_hello:
QuicVersion negotiated_version;
QuicVersion initial_version;
case encrypted_extensions:
QuicVersion supported_versions<2..2^8-4>;
};
TransportParameter parameters<30..2^16-1>;
} TransportParameters;
]]>The extension_data field of the quic_transport_parameters extension defined in
contains a TransportParameters value. TLS encoding rules are
therefore used to encode the transport parameters.QUIC encodes transport parameters into a sequence of octets, which are then
included in the cryptographic handshake. Once the handshake completes, the
transport parameters declared by the peer are available. Each endpoint
validates the value provided by its peer. In particular, version negotiation
MUST be validated (see ) before the connection
establishment is considered properly complete.Definitions for each of the defined transport parameters are included in
.An endpoint MUST include the following parameters in its encoded
TransportParameters:
The initial stream level flow control offset parameter is encoded as an
unsigned 32-bit integer in units of octets. The sender of this parameter
indicates that the flow control offset for all stream data sent toward it is
this value.
The connection level flow control offset parameter contains the initial
connection flow control window encoded as an unsigned 32-bit integer in units
of 1024 octets. That is, the value here is multiplied by 1024 to determine
the actual flow control offset. The sender of this parameter sets the byte
offset for connection level flow control to this value. This is equivalent to
sending a WINDOW_UPDATE () for the connection
immediately after completing the handshake.
The maximum number of concurrent streams parameter is encoded as an unsigned
32-bit integer.
The idle timeout is a value in seconds that is encoded as an unsigned 16-bit
integer. The maximum value is 600 seconds (10 minutes).An endpoint MAY use the following transport parameters:
The truncated connection identifier parameter indicates that packets sent to
the peer can omit the connection ID. This can be used by an endpoint where
the 5-tuple is sufficient to identify a connection. This parameter is zero
length. Omitting the parameter indicates that the endpoint relies on the
connection ID being present in every packet.Transport parameters from the server SHOULD be remembered by the client for use
with 0-RTT data. A client that doesn’t remember values from a previous
connection can instead assume the following values: stream_fc_offset (65535),
connection_fc_offset (65535), concurrent_streams (10), idle_timeout (600),
truncate_connection_id (absent).If assumed values change as a result of completing the handshake, the client is
expected to respect the new values. This introduces some potential problems,
particularly with respect to transport parameters that establish limits:A client might exceed a newly declared connection or stream flow control limit
with 0-RTT data. If this occurs, the client ceases transmission as though the
flow control limit was reached. Once WINDOW_UPDATE frames indicating an
increase to the affected flow control offsets is received, the client can
recommence sending.Similarly, a client might exceed the concurrent stream limit declared by the
server. A client MUST reset any streams that exceed this limit. A server
SHOULD reset any streams it cannot handle with a code that allows the client
to retry any application action bound to those streams.A server MAY close a connection if remembered or assumed 0-RTT transport
parameters cannot be supported, using an error code that is appropriate to the
specific condition. For example, a QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA
might be used to indicate that exceeding flow control limits caused the error.
A client that has a connection closed due to an error condition SHOULD NOT
attempt 0-RTT when attempting to create a new connection.New transport parameters can be used to negotiate new protocol behavior. An
endpoint MUST ignore transport parameters that it does not support. Absence of
a transport parameter therefore disables any optional protocol feature that is
negotiated using the parameter.The definition of a transport parameter SHOULD include a default value that a
client can use when establishing a new connection. If no default is specified,
the value can be assumed to be absent when attempting 0-RTT.New transport parameters can be registered according to the rules in
.The transport parameters include three fields that encode version information.
These retroactively authenticate the version negotiation (see
) that is performed prior to the cryptographic handshake.The cryptographic handshake provides integrity protection for the negotiated
version as part of the transport parameters (see ). As
a result, modification of version negotiation packets by an attacker can be
detected.The client includes two fields in the transport parameters:The negotiated_version is the version that was finally selected for use. This
MUST be identical to the value that is on the packet that carries the
ClientHello. A server that receives a negotiated_version that does not match
the version of QUIC that is in use MUST terminate the connection with a
QUIC_VERSION_NEGOTIATION_MISMATCH error code.The initial_version is the version that the client initially attempted to use.
If the server did not send a version negotiation packet ,
this will be identical to the negotiated_version.A server that processes all packets in a stateful fashion can remember how
version negotiation was performed and validate the initial_version value.A server that does not maintain state for every packet it receives (i.e., a
stateless server) uses a different process. If the initial and negotiated
versions are the same, a stateless server can accept the value.If the initial version is different from the negotiated_version, a stateless
server MUST check that it would have sent a version negotiation packet if it had
received a packet with the indicated initial_version. If a server would have
accepted the version included in the initial_version and the value differs from
the value of negotiated_version, the server MUST terminate the connection with a
QUIC_VERSION_NEGOTIATION_MISMATCH error.The server includes a list of versions that it would send in any version
negotiation packet () in supported_versions. This value is
set even if it did not send a version negotiation packet.The client can validate that the negotiated_version is included in the
supported_versions list and - if version negotiation was performed - that it
would have selected the negotiated version. A client MUST terminate the
connection with a QUIC_VERSION_NEGOTIATION_MISMATCH error code if the
negotiated_version value is not included in the supported_versions list. A
client MUST terminate with a QUIC_VERSION_NEGOTIATION_MISMATCH error code if
version negotiation occurred but it would have selected a different version
based on the value of the supported_versions list.Transport protocols commonly spend a round trip checking that a client owns the
transport address (IP and port) that it claims. Verifying that a client can
receive packets sent to its claimed transport address protects against spoofing
of this information by malicious clients.This technique is used primarily to avoid QUIC from being used for traffic
amplification attack. In such an attack, a packet is sent to a server with
spoofed source address information that identifies a victim. If a server
generates more or larger packets in response to that packet, the attacker can
use the server to send more data toward the victim than it would be able to send
on its own.Several methods are used in QUIC to mitigate this attack. Firstly, the initial
handshake packet from a client is padded to at least 1280 octets. This allows a
server to send a similar amount of data without risking causing an amplication
attack toward an unproven remote address.A server eventually confirms that a client has received its messages when the
cryptographic handshake successfully completes. This might be insufficient,
either because the server wishes to avoid the computational cost of completing
the handshake, or it might be that the size of the packets that are sent during
the handshake is too large. This is especially important for 0-RTT, where the
server might wish to provide application data traffic - such as a response to a
request - in response to the data carried in the early data from the client.To send additional data prior to completing the cryptographic handshake, the
server then needs to validate that the client owns the address that it claims.Source address validation is therefore performed during the establishment of a
connection. TLS provides the tools that support the feature, but basic
validation is performed by the core transport protocol.QUIC uses token-based address validation. Any time the server wishes to
validate a client address, it provides the client with a token. As long as the
token cannot be easily guessed (see ), if the client is able
to return that token, it proves to the server that it received the token.During the processing of the cryptographic handshake messages from a client, TLS
will request that QUIC make a decision about whether to proceed based on the
information it has. TLS will provide QUIC with any token that was provided by
the client. For an initial packet, QUIC can decide to abort the connection,
allow it to proceed, or request address validation.If QUIC decides to request address validation, it provides the cryptographic
handshake with a token. The contents of this token are consumed by the server
that generates the token, so there is no need for a single well-defined format.
A token could include information about the claimed client address (IP and
port), a timestamp, and any other supplementary information the server will need
to validate the token in the future.The cryptographic handshake is responsible for enacting validation by sending
the address validation token to the client. A legitimate client will include a
copy of the token when it attempts to continue the handshake. The cryptographic
handshake extracts the token then asks QUIC a second time whether the token is
acceptable. In response, QUIC can either abort the connection or permit it to
proceed.A connection MAY be accepted without address validation - or with only limited
validation - but a server SHOULD limit the data it sends toward an unvalidated
address. Successful completion of the cryptographic handshake implicitly
provides proof that the client has received packets from the server.A server MAY provide clients with an address validation token during one
connection that can be used on a subsequent connection. Address validation is
especially important with 0-RTT because a server potentially sends a significant
amount of data to a client in response to 0-RTT data.A different type of token is needed when resuming. Unlike the token that is
created during a handshake, there might be some time between when the token is
created and when the token is subsequently used. Thus, a resumption token
SHOULD include an expiration time. It is also unlikely that the client port
number is the same on two different connections; validating the port is
therefore unlikely to be successful.This token can be provided to the cryptographic handshake immediately after
establishing a connection. QUIC might also generate an updated token if
significant time passes or the client address changes for any reason (see
). The cryptographic handshake is responsible for providing the
client with the token. In TLS the token is included in the ticket that is used
for resumption and 0-RTT, which is carried in a NewSessionTicket message.An address validation token MUST be difficult to guess. Including a large
enough random value in the token would be sufficient, but this depends on the
server remembering the value it sends to clients.A token-based scheme allows the server to offload any state associated with
validation to the client. For this design to work, the token MUST be covered by
integrity protection against modification or falsification by clients. Without
integrity protection, malicious clients could generate or guess values for
tokens that would be accepted by the server. Only the server requires access to
the integrity protection key for tokens.In TLS the address validation token is often bundled with the information that
TLS requires, such as the resumption secret. In this case, adding integrity
protection can be delegated to the cryptographic handshake protocol, avoiding
redundant protection. If integrity protection is delegated to the cryptographic
handshake, an integrity failure will result in immediate cryptographic handshake
failure. If integrity protection is performed by QUIC, QUIC MUST abort the
connection if the integrity check fails with a QUIC_ADDRESS_VALIDATION_FAILURE
error code.QUIC connections are identified by their 64-bit Connection ID. QUIC’s
consistent connection ID allows connections to survive changes to the
client’s IP and/or port, such as those caused by client or server
migrating to a new network. QUIC also provides automatic
cryptographic verification of a client which has changed its IP
address because the client continues to use the same session key for
encrypting and decrypting packets.DISCUSS: Simultaneous migration. Is this reasonable?TODO: Perhaps move mitigation techniques from Security Considerations here.Connections should remain open until they become idle for a pre-negotiated
period of time. A QUIC connection, once established, can be terminated in one
of three ways:Explicit Shutdown: An endpoint sends a CONNECTION_CLOSE frame to
initiate a connection termination. An endpoint may send a GOAWAY frame to
the peer prior to a CONNECTION_CLOSE to indicate that the connection will
soon be terminated. A GOAWAY frame signals to the peer that any active
streams will continue to be processed, but the sender of the GOAWAY will not
initiate any additional streams and will not accept any new incoming streams.
On termination of the active streams, a CONNECTION_CLOSE may be sent. If an
endpoint sends a CONNECTION_CLOSE frame while unterminated streams are active
(no FIN bit or RST_STREAM frames have been sent or received for one or more
streams), then the peer must assume that the streams were incomplete and were
abnormally terminated.Implicit Shutdown: The default idle timeout for a QUIC connection is 30
seconds, and is a required parameter in connection negotiation. The maximum
is 10 minutes. If there is no network activity for the duration of the idle
timeout, the connection is closed. By default a CONNECTION_CLOSE frame will
be sent. A silent close option can be enabled when it is expensive to send
an explicit close, such as mobile networks that must wake up the radio.Abrupt Shutdown: An endpoint may send a Public Reset packet at any time
during the connection to abruptly terminate an active connection. A Public
Reset packet SHOULD only be used as a final recourse. Commonly, a public
reset is expected to be sent when a packet on an established connection is
received by an endpoint that is unable decrypt the packet. For instance, if
a server reboots mid-connection and loses any cryptographic state associated
with open connections, and then receives a packet on an open connection, it
should send a Public Reset packet in return. (TODO: articulate rules around
when a public reset should be sent.)TODO: Connections that are terminated are added to a TIME_WAIT list at the
server, so as to absorb any straggler packets in the network. Discuss TIME_WAIT
list.As described in , Regular packets contain one or more frames.
We now describe the various QUIC frame types that can be present in a Regular
packet. The use of these frames and various frame header bits are described in
subsequent sections.STREAM frames implicitly create a stream and carry stream data. The type byte
for a STREAM frame contains embedded flags, and is formatted as 1FDOOOSS.
These bits are parsed as follows:The leftmost bit must be set to 1, indicating that this is a STREAM frame.F is the FIN bit, which is used for stream termination.The D bit indicates whether a Data Length field is present in the STREAM
header. When set to 0, this field indicates that the Stream Data field
extends to the end of the packet. When set to 1, this field indicates that
Data Length field contains the length (in bytes) of the Stream Data field.
The option to omit the length should only be used when the packet is a
“full-sized” packet, to avoid the risk of corruption via padding.The OOO bits encode the length of the Offset header field as 0, 16, 24,
32, 40, 48, 56, or 64 bits long.The SS bits encode the length of the Stream ID header field as 8, 16, 24,
or 32 bits. (DISCUSS: Consider making this 8, 16, 32, 64.)A STREAM frame is shown below.The STREAM frame contains the following fields:
An optional 16-bit unsigned number specifying the length of the Stream Data
field in this STREAM frame. This field is present when the D bit is set to
1.
A variable-sized unsigned ID unique to this stream.
A variable-sized unsigned number specifying the byte offset in the stream for
the data in this STREAM frame. The first byte in the stream has an offset of
0. The largest offset delivered on a stream - the sum of the re-constructed
offset and data length - MUST be less than 2^64.
The bytes from the designated stream to be delivered.A STREAM frame MUST have either non-zero data length or the FIN bit set.Stream multiplexing is achieved by interleaving STREAM frames from multiple
streams into one or more QUIC packets. A single QUIC packet MAY bundle STREAM
frames from multiple streams.Implementation note: One of the benefits of QUIC is avoidance of head-of-line
blocking across multiple streams. When a packet loss occurs, only streams with
data in that packet are blocked waiting for a retransmission to be received,
while other streams can continue making progress. Note that when data from
multiple streams is bundled into a single QUIC packet, loss of that packet
blocks all those streams from making progress. An implementation is therefore
advised to bundle as few streams as necessary in outgoing packets without losing
transmission efficiency to underfilled packets.Receivers send ACK frames to inform senders which packets they have received and
processed, as well as which packets are considered missing. The ACK frame
contains between 1 and 256 ACK blocks. ACK blocks are ranges of acknowledged
packets.To limit ACK blocks to those that have not yet been received by the sender, the
receiver SHOULD track which ACK frames have been acknowledged by its peer. Once
an ACK frame has been acknowledged, the packets it acknowledges SHOULD not be
acknowledged again. To handle cases where the receiver is only sending ACK
frames, and hence will not receive acknowledgments for its packets, it MAY send
a PING frame at most once per RTT to explicitly request acknowledgment.To limit receiver state or the size of ACK frames, a receiver MAY limit the
number of ACK blocks it sends. A receiver can do this even without receiving
acknowledgment of its ACK frames, with the knowledge this could cause the sender
to unnecessarily retransmit some data.Unlike TCP SACKs, QUIC ACK blocks are cumulative and therefore irrevocable.
Once a packet has been acknowledged, even if it does not appear in a future ACK
frame, it is assumed to be acknowledged.QUIC ACK frames contain a timestamp section with up to 255 timestamps.
Timestamps enable better congestion control, but are not required for correct
loss recovery, and old timestamps are less valuable, so it is not guaranteed
every timestamp will be received by the sender. A receiver SHOULD send a
timestamp exactly once for each received packet containing retransmittable
frames. A receiver MAY send timestamps for non-retransmittable packets.A sender MAY intentionally skip packet numbers to introduce entropy into the
connection, to avoid opportunistic acknowledgement attacks. The sender MUST
close the connection if an unsent packet number is acknowledged. The format of
the ACK frame is efficient at expressing blocks of missing packets; skipping
packet numbers between 1 and 255 effectively provides up to 8 bits of efficient
entropy on demand, which should be adequate protection against most
opportunistic acknowledgement attacks.The type byte for a ACK frame contains embedded flags, and is formatted as
01NULLMM. These bits are parsed as follows:The first two bits must be set to 01 indicating that this is an ACK frame.The N bit indicates whether the frame has more than 1 range of acknowledged
packets (i.e., whether the ACK Block Section contains a Num Blocks field).The U bit is unused and MUST be set to zero.The two LL bits encode the length of the Largest Acknowledged field as 1, 2,
4, or 6 bytes long.The two MM bits encode the length of the ACK Block Length fields as 1, 2,
4, or 6 bytes long.An ACK frame is shown below.The fields in the ACK frame are as follows:
An optional 8-bit unsigned value specifying the number of additional ACK
blocks (besides the required First ACK Block) in this ACK frame. Only present
if the ‘N’ flag bit is 1.
An unsigned 8-bit number specifying the total number of <packet number,
timestamp> pairs in the Timestamp Section.
A variable-sized unsigned value representing the largest packet number the
peer is acknowledging in this packet (typically the largest that the peer has
seen thus far.)
The time from when the largest acknowledged packet, as indicated in the
Largest Acknowledged field, was received by this peer to when this ACK was
sent.
Contains one or more blocks of packet numbers which have been successfully
received, see .
Contains zero or more timestamps reporting transit delay of received packets.
See .The ACK Block Section contains between one and 256 blocks of packet numbers
which have been successfully received. If the Num Blocks field is absent, only
the First ACK Block length is present in this section. Otherwise, the Num Blocks
field indicates how many additional blocks follow the First ACK Block Length
field.The fields in the ACK Block Section are:
An unsigned packet number delta that indicates the number of contiguous
additional packets being acknowledged starting at the Largest Acknowledged.
An unsigned number specifying the number of contiguous missing packets from
the end of the previous ACK block to the start of the next. Repeated “Num
Blocks” times.
An unsigned packet number delta that indicates the number of contiguous
packets being acknowledged starting after the end of the previous gap.
Repeated “Num Blocks” times.The Timestamp Section contains between zero and 255 measurements of packet
receive times relative to the beginning of the connection.The fields in the Timestamp Section are:
An optional 8-bit unsigned packet number delta specifying the delta between
the largest acknowledged and the first packet whose timestamp is being
reported. In other words, this first packet number may be computed as
(Largest Acknowledged - Delta Largest Acknowledged.)
An optional 32-bit unsigned value specifying the time delta in microseconds,
from the beginning of the connection to the arrival of the packet indicated by
Delta Largest Acknowledged.
This field has the same semantics and format as “Delta Largest Acknowledged”.
Repeated “Num Timestamps - 1” times.
An optional 16-bit unsigned value specifying time delta from the previous
reported timestamp. It is encoded in the same format as the ACK Delay.
Repeated “Num Timestamps - 1” times.The timestamp section lists packet receipt timestamps ordered by timestamp.DISCUSS_AND_REPLACE: Perhaps make this format simpler.The time format used in the ACK frame above is a 16-bit unsigned float with 11
explicit bits of mantissa and 5 bits of explicit exponent, specifying time in
microseconds. The bit format is loosely modeled after IEEE 754. For example, 1
microsecond is represented as 0x1, which has an exponent of zero, presented in
the 5 high order bits, and mantissa of 1, presented in the 11 low order bits.
When the explicit exponent is greater than zero, an implicit high-order 12th bit
of 1 is assumed in the mantissa. For example, a floating value of 0x800 has an
explicit exponent of 1, as well as an explicit mantissa of 0, but then has an
effective mantissa of 4096 (12th bit is assumed to be 1). Additionally, the
actual exponent is one-less than the explicit exponent, and the value represents
4096 microseconds. Any values larger than the representable range are clamped
to 0xFFFF.ACK frames that acknowledge protected packets MUST be carried in a packet that
has an equivalent or greater level of packet protection.Packets that are protected with 1-RTT keys MUST be acknowledged in packets that
are also protected with 1-RTT keys.A packet that is not protected and claims to acknowledge a packet number that
was sent with packet protection is not valid. An unprotected packet that
carries acknowledgments for protected packets MUST be discarded in its entirety.Packets that a client sends with 0-RTT packet protection MUST be acknowledged by
the server in packets protected by 1-RTT keys. This can mean that the client is
unable to use these acknowledgments if the server cryptographic handshake
messages are delayed or lost. Note that the same limitation applies to other
data sent by the server protected by the 1-RTT keys.Unprotected packets, such as those that carry the initial cryptographic
handshake messages, MAY be acknowledged in unprotected packets. Unprotected
packets are vulnerable to falsification or modification. Unprotected packets
can be acknowledged along with protected packets in a protected packet.An endpoint SHOULD acknowledge packets containing cryptographic handshake
messages in the next unprotected packet that it sends, unless it is able to
acknowledge those packets in later packets protected by 1-RTT keys. At the
completion of the cryptographic handshake, both peers send unprotected packets
containing cryptographic handshake messages followed by packets protected by
1-RTT keys. An endpoint SHOULD acknowledge the unprotected packets that complete
the cryptographic handshake in a protected packet, because its peer is
guaranteed to have access to 1-RTT packet protection keys.For instance, a server acknowledges a TLS ClientHello in the packet that carries
the TLS ServerHello; similarly, a client can acknowledge a TLS HelloRetryRequest
in the packet containing a second TLS ClientHello. The complete set of server
handshake messages (TLS ServerHello through to Finished) might be acknowledged
by a client in protected packets, because it is certain that the server is able
to decipher the packet.The WINDOW_UPDATE frame (type=0x04) informs the peer of an increase in an
endpoint’s flow control receive window for either a single stream, or the entire
connection as a whole.The frame is as follows:The fields in the WINDOW_UPDATE frame are as follows:
ID of the stream whose flow control windows is being updated, or 0 to specify
the connection-level flow control window.
A 64-bit unsigned integer indicating the flow control offset for the given
stream (for a stream ID other than 0) or the entire connection.The flow control offset is expressed in units of octets for individual streams
(for stream identifiers other than 0).The connection-level flow control offset is expressed in units of 1024 octets
(for a stream identifier of 0). That is, the connection-level flow control
offset is determined by multiplying the encoded value by 1024.An endpoint accounts for the maximum offset of data that is sent or received on
a stream. Loss or reordering can mean that the maximum offset is greater than
the total size of data received on a stream. Similarly, receiving STREAM frames
might not increase the maximum offset on a stream. A STREAM frame with a FIN
bit set or RST_STREAM causes the final offset for a stream to be fixed.The maximum data offset on a stream MUST NOT exceed the stream flow control
offset advertised by the receiver. The sum of the maximum data offsets of all
streams (including closed streams) MUST NOT exceed the connection flow control
offset advertised by the receiver. An endpoint MUST terminate a connection with
a QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA error if it receives more data than
the largest flow control offset that it has sent, unless this is a result of a
change in the initial offsets (see ).A sender sends a BLOCKED frame (type=0x05) when it is ready to send data (and
has data to send), but is currently flow control blocked. BLOCKED frames are
purely informational frames, but extremely useful for debugging purposes. A
receiver of a BLOCKED frame should simply discard it (after possibly printing a
helpful log message). The frame is as follows:The BLOCKED frame contains a single field:
A 32-bit unsigned number indicating the stream which is flow control blocked.
A non-zero Stream ID field specifies the stream that is flow control blocked.
When zero, the Stream ID field indicates that the connection is flow control
blocked.An endpoint may use a RST_STREAM frame (type=0x01) to abruptly terminate a
stream. The frame is as follows:The fields are:
A 32-bit error code which indicates why the stream is being closed.
The 32-bit Stream ID of the stream being terminated.
A 64-bit unsigned integer indicating the absolute byte offset of the end of
data written on this stream by the RST_STREAM sender.The PADDING frame (type=0x00) has no semantic value. PADDING frames can be used
to increase the size of a packet. Padding can be used to increase an initial
client packet to the minimum required size, or to provide protection against
traffic analysis for protected packets.A PADDING frame has no content. That is, a PADDING frame consists of the single
octet that identifies the frame as a PADDING frame.Endpoints can use PING frames (type=0x07) to verify that their peers are still
alive or to check reachability to the peer. The PING frame contains no
additional fields. The receiver of a PING frame simply needs to acknowledge the
packet containing this frame. The PING frame SHOULD be used to keep a connection
alive when a stream is open. The default is to send a PING frame after 15
seconds of quiescence. A PING frame has no additional fields.An endpoint sends a CONNECTION_CLOSE frame (type=0x02) to notify its peer that
the connection is being closed. If there are open streams that haven’t been
explicitly closed, they are implicitly closed when the connection is closed.
(Ideally, a GOAWAY frame would be sent with enough time that all streams are
torn down.) The frame is as follows:The fields of a CONNECTION_CLOSE frame are as follows:
A 32-bit error code which indicates the reason for closing this connection.
A 16-bit unsigned number specifying the length of the reason phrase. This may
be zero if the sender chooses to not give details beyond the Error Code.
An optional human-readable explanation for why the connection was closed.An endpoint uses a GOAWAY frame (type=0x03) to initiate a graceful shutdown of a
connection. The endpoints will continue to use any active streams, but the
sender of the GOAWAY will not initiate or accept any additional streams beyond
those indicated. The GOAWAY frame is as follows:The fields of a GOAWAY frame are:
The highest-numbered, client-initiated stream on which the endpoint sending
the GOAWAY frame either sent data, or received and delivered data. All
higher-numbered, client-initiated streams (that is, odd-numbered streams) are
implicitly reset by sending or receiving the GOAWAY frame.
The highest-numbered, server-initiated stream on which the endpoint sending
the GOAWAY frame either sent data, or received and delivered data. All
higher-numbered, server-initiated streams (that is, even-numbered streams) are
implicitly reset by sending or receiving the GOAWAY frame.A GOAWAY frame indicates that any application layer actions on streams with
higher numbers than those indicated can be safely retried because no data was
exchanged. An endpoint MUST set the value of the Largest Client or Server
Stream ID to be at least as high as the highest-numbered stream on which it
either sent data or received and delivered data to the application protocol that
uses QUIC.An endpoint MAY choose a larger stream identifier if it wishes to allow for a
number of streams to be created. This is especially valuable for peer-initiated
streams where packets creating new streams could be in transit; using a larger
stream number allows those streams to complete.In addition to initiating a graceful shutdown of a connection, GOAWAY MAY be
sent immediately prior to sending a CONNECTION_CLOSE frame that is sent as a
result of detecting a fatal error. Higher-numbered streams than those indicated
in the GOAWAY frame can then be retried.The Path Maximum Transmission Unit (PTMU) is the maximum size of the entire IP
header, UDP header, and UDP payload. The UDP payload includes the QUIC public
header, encrypted payload, and any authentication fields.All QUIC packets SHOULD be sized to fit within the estimated PMTU to avoid IP
fragmentation or packet drops. To optimize bandwidth efficiency, endpoints
SHOULD use Packetization Layer PMTU Discovery () and MAY use PMTU
Discovery (, ) for detecting the PMTU, setting the PMTU
appropriately, and storing the result of previous PMTU determinations.In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP packets
larger than 1280 octets. Assuming the minimum IP header size, this results in
a UDP payload length of 1232 octets for IPv6 and 1252 octets for IPv4.QUIC endpoints that implement any kind of PMTU discovery SHOULD maintain an
estimate for each combination of local and remote IP addresses (as each pairing
could have a different maximum MTU in the path).QUIC depends on the network path supporting a MTU of at least 1280 octets. This
is the IPv6 minimum and therefore also supported by most modern IPv4 networks.
An endpoint MUST NOT reduce their MTU below this number, even if it receives
signals that indicate a smaller limit might exist.Clients MUST ensure that the first packet in a connection, and any
retransmissions of those octets, has a total size (including IP and UDP headers)
of at least 1280 bytes. This might require inclusion of PADDING frames. It is
RECOMMENDED that a packet be padded to exactly 1280 octets unless the client has
a reasonable assurance that the PMTU is larger. Sending a packet of this size
ensures that the network path supports an MTU of this size and helps mitigate
amplification attacks caused by server responses toward an unverified client
address.Servers MUST reject the first plaintext packet received from a client if it its
total size is less than 1280 octets, to mitigate amplification attacks.If a QUIC endpoint determines that the PMTU between any pair of local and remote
IP addresses has fallen below 1280 octets, it MUST immediately cease sending
QUIC packets between those IP addresses. This may result in abrupt termination
of the connection if all pairs are affected. In this case, an endpoint SHOULD
send a Public Reset packet to indicate the failure. The application SHOULD
attempt to use TLS over TCP instead.A sender bundles one or more frames in a Regular QUIC packet (see ).A sender SHOULD minimize per-packet bandwidth and computational costs by
bundling as many frames as possible within a QUIC packet. A sender MAY wait for
a short period of time to bundle multiple frames before sending a packet that is
not maximally packed, to avoid sending out large numbers of small packets. An
implementation may use heuristics about expected application sending behavior to
determine whether and for how long to wait. This waiting period is an
implementation decision, and an implementation should be careful to delay
conservatively, since any delay is likely to increase application-visible
latency.Regular QUIC packets are “containers” of frames; a packet is never retransmitted
whole. How an endpoint handles the loss of the frame depends on the type of the
frame. Some frames are simply retransmitted, some have their contents moved to
new frames, and others are never retransmitted.When a packet is detected as lost, the sender re-sends any frames as necessary:All application data sent in STREAM frames MUST be retransmitted, unless the
endpoint has sent a RST_STREAM for that stream. When an endpoint sends a
RST_STREAM frame, data outstanding on that stream SHOULD NOT be retransmitted,
since subsequent data on this stream is expected to not be delivered by the
receiver.ACK and PADDING frames MUST NOT be retransmitted. ACK frames are cumulative,
so new frames containing updated information will be sent as described in
.All other frames MUST be retransmitted.Upon detecting losses, a sender MUST take appropriate congestion control action.
The details of loss detection and congestion control are described in
.A packet MUST NOT be acknowledged until packet protection has been successfully
removed and all frames contained in the packet have been processed. For STREAM
frames, this means the data has been queued (but not necessarily delivered to
the application). This also means that any stream state transitions triggered
by STREAM or RST_STREAM frames have occurred. Once the packet has been fully
processed, a receiver acknowledges receipt by sending one or more ACK frames
containing the packet number of the received packet.To avoid creating an indefinite feedback loop, an endpoint MUST NOT generate an
ACK frame in response to a packet containing only ACK or PADDING frames.Strategies and implications of the frequency of generating acknowledgments are
discussed in more detail in .Traditional ICMP-based path MTU discovery in IPv4 ( is potentially
vulnerable to off-path attacks that successfully guess the IP/port 4-tuple and
reduce the MTU to a bandwidth-inefficient value. TCP connections mitigate this
risk by using the (at minimum) 8 bytes of transport header echoed in the ICMP
message to validate the TCP sequence number as valid for the current
connection. However, as QUIC operates over UDP, in IPv4 the echoed information
could consist only of the IP and UDP headers, which usually has insufficient
entropy to mitigate off-path attacks.As a result, endpoints that implement PMTUD in IPv4 SHOULD take steps to
mitigate this risk. For instance, an application could:Set the IPv4 Don’t Fragment (DF) bit on a small proportion of packets, so that
most invalid ICMP messages arrive when there are no DF packets outstanding, and
can therefore be identified as spurious.Store additional information from the IP or UDP headers from DF packets (for
example, the IP ID or UDP checksum) to further authenticate incoming Datagram
Too Big messages.Any reduction in PMTU due to a report contained in an ICMP packet is
provisional until QUIC’s loss detection algorithm determines that the packet is
actually lost.Streams in QUIC provide a lightweight, ordered, and bidirectional byte-stream
abstraction modeled closely on HTTP/2 streams .Streams can be created either by the client or the server, can concurrently send
data interleaved with other streams, and can be cancelled.Data that is received on a stream is delivered in order within that stream, but
there is no particular delivery order across streams. Transmit ordering among
streams is left to the implementation.The creation and destruction of streams are expected to have minimal bandwidth
and computational cost. A single STREAM frame may create, carry data for, and
terminate a stream, or a stream may last the entire duration of a connection.Streams are individually flow controlled, allowing an endpoint to limit memory
commitment and to apply back pressure.An alternative view of QUIC streams is as an elastic “message” abstraction,
similar to the way ephemeral streams are used in SST , which may be a
more appealing description for some applications.The semantics of QUIC streams is based on HTTP/2 streams, and the lifecycle of a
QUIC stream therefore closely follows that of an HTTP/2 stream ,
with some differences to accommodate the possibility of out-of-order delivery
due to the use of multiple streams in QUIC. The lifecycle of a QUIC stream is
shown in the following figure and described below.| |Note that this diagram shows stream state transitions and the frames and flags
that affect those transitions only. For the purpose of state transitions, the
FIN flag is processed as a separate event to the frame that bears it; a STREAM
frame with the FIN flag set can cause two state transitions. When the FIN flag
is sent on an empty STREAM frame, the offset in the STREAM frame MUST be one
greater than the last data byte sent on this stream.The recipient of a frame which changes stream state will have a delayed view of
the state of a stream while the frame is in transit. Endpoints do not
coordinate the creation of streams; they are created unilaterally by either
endpoint. The negative consequences of a mismatch in states are limited to the
“closed” state after sending RST_STREAM, where frames might be received for some
time after closing. Endpoints can use acknowledgments to understand the peer’s
subjective view of stream state at any given time.Streams have the following states:All streams start in the “idle” state.The following transitions are valid from this state:Sending or receiving a STREAM frame causes the stream to become “open”. The
stream identifier is selected as described in . The same
STREAM frame can also cause a stream to immediately become “half-closed”.Receiving a STREAM frame on a peer-initiated stream (that is, a packet sent by a
server on an even-numbered stream or a client packet on an odd-numbered stream)
also causes all lower-numbered “idle” streams in the same direction to become
“open”. This could occur if a peer begins sending on streams in a different
order to their creation, or it could happen if packets are lost or reordered in
transit.Receiving any frame other than STREAM or RST_STREAM on a stream in this state
MUST be treated as a connection error () of type YYYY.A stream in the “open” state may be used by both peers to send frames of any
type. In this state, a sending peer must observe the flow-control limit
advertised by its receiving peer ().From this state, either endpoint can send a frame with the FIN flag set, which
causes the stream to transition into one of the “half-closed” states. An
endpoint sending an FIN flag causes the stream state to become “half-closed
(local)”. An endpoint receiving a FIN flag causes the stream state to become
“half-closed (remote)” once all preceding data has arrived. The receiving
endpoint MUST NOT consider the stream state to have changed until all data has
arrived.Either endpoint can send a RST_STREAM frame from this state, causing it to
transition immediately to “closed”.A stream that is in the “half-closed (local)” state MUST NOT be used for sending
STREAM frames; WINDOW_UPDATE and RST_STREAM MAY be sent in this state.A stream transitions from this state to “closed” when a STREAM frame that
contains a FIN flag is received and all prior data has arrived, or when either
peer sends a RST_STREAM frame.An endpoint that closes a stream MUST NOT send data beyond the final offset that
it has chosen, see for details.An endpoint can receive any type of frame in this state. Providing flow-control
credit using WINDOW_UPDATE frames is necessary to continue receiving
flow-controlled frames. In this state, a receiver MAY ignore WINDOW_UPDATE
frames for this stream, which might arrive for a short period after a frame
bearing the FIN flag is sent.A stream that is “half-closed (remote)” is no longer being used by the peer to
send any data. In this state, a sender is no longer obligated to maintain a
receiver stream-level flow-control window.A stream that is in the “half-closed (remote)” state will have a final offset
for received data, see for details.A stream in this state can be used by the endpoint to send frames of any type.
In this state, the endpoint continues to observe advertised stream-level and
connection-level flow-control limits ().A stream can transition from this state to “closed” by sending a frame that
contains a FIN flag or when either peer sends a RST_STREAM frame.The “closed” state is the terminal state.An endpoint will learn the final offset of the data it receives on a stream when
it enters the “half-closed (remote)” or “closed” state. The final offset is
carried explicitly in the RST_STREAM frame; otherwise, the final offset is the
offset of the end of the data carried in STREAM frame marked with a FIN flag.An endpoint MUST NOT send data on a stream at or beyond the final offset.Once a final offset for a stream is known, it cannot change. If a RST_STREAM or
STREAM frame causes the final offset to change for a stream, an endpoint SHOULD
respond with a QUIC_STREAM_DATA_AFTER_TERMINATION error (see
). A receiver SHOULD treat receipt of data at or beyond the
final offset as a QUIC_STREAM_DATA_AFTER_TERMINATION error. Generating these
errors is not mandatory, but only because requiring that an endpoint generate
these errors also means that the endpoint needs to maintain the final offset
state for closed streams, which could mean a significant state commitment.An endpoint that receives a RST_STREAM frame (and which has not sent a FIN or a
RST_STREAM) MUST immediately respond with a RST_STREAM frame, and MUST NOT send
any more data on the stream. This endpoint may continue receiving frames for
the stream on which a RST_STREAM is received.If this state is reached as a result of sending a RST_STREAM frame, the peer
that receives the RST_STREAM frame might have already sent – or enqueued for
sending – frames on the stream that cannot be withdrawn. An endpoint MUST
ignore frames that it receives on closed streams after it has sent a RST_STREAM
frame. An endpoint MAY choose to limit the period over which it ignores frames
and treat frames that arrive after this time as being in error.STREAM frames received after sending RST_STREAM are counted toward the
connection and stream flow-control windows. Even though these frames might be
ignored, because they are sent before their sender receives the RST_STREAM, the
sender will consider the frames to count against its flow-control windows.In the absence of more specific guidance elsewhere in this document,
implementations SHOULD treat the receipt of a frame that is not expressly
permitted in the description of a state as a connection error
(). Frames of unknown types are ignored.(TODO: QUIC_STREAM_NO_ERROR is a special case. Write it up.)Streams are identified by an unsigned 32-bit integer, referred to as the
StreamID. To avoid StreamID collision, clients MUST initiate streams usinge
odd-numbered StreamIDs; streams initiated by the server MUST use even-numbered
StreamIDs.A StreamID of zero (0x0) is reserved and used for connection-level flow control
frames (); the StreamID of zero cannot be used to establish a
new stream.StreamID 1 (0x1) is reserved for the cryptographic handshake. StreamID 1 MUST
NOT be used for application data, and MUST be the first client-initiated stream.A QUIC endpoint cannot reuse a StreamID on a given connection. Streams MUST be
created in sequential order. Open streams can be used in any order. Streams
that are used out of order result in lower-numbered streams in the same
direction being counted as open.All streams, including stream 1, count toward this limit. Thus, a concurrent
stream limit of 0 will cause a connection to be unusable. Application protocols
that use QUIC might require a certain minimum number of streams to function
correctly. If a peer advertises an concurrent stream limit (concurrent_streams)
that is too small for the selected application protocol to function, an endpoint
MUST terminate the connection with an error of type QUIC_TOO_MANY_OPEN_STREAMS
().An endpoint limits the number of concurrently active incoming streams by setting
the concurrent stream limit (see ) in the
transport parameters. The maximum concurrent streams setting is specific to each
endpoint and applies only to the peer that receives the setting. That is,
clients specify the maximum number of concurrent streams the server can
initiate, and servers specify the maximum number of concurrent streams the
client can initiate.Streams that are in the “open” state or in either of the “half-closed” states
count toward the maximum number of streams that an endpoint is permitted to
open. Streams in any of these three states count toward the limit advertised in
the concurrent stream limit.A recently closed stream MUST also be considered to count toward this limit
until packets containing all frames required to close the stream have been
acknowledged. For a stream which closed cleanly, this means all STREAM frames
have been acknowledged; for a stream which closed abruptly, this means the
RST_STREAM frame has been acknowledged.Endpoints MUST NOT exceed the limit set by their peer. An endpoint that
receives a STREAM frame that causes its advertised concurrent stream limit to be
exceeded MUST treat this as a stream error of type QUIC_TOO_MANY_OPEN_STREAMS
().Once a stream is created, endpoints may use the stream to send and receive data.
Each endpoint may send a series of STREAM frames encapsulating data on a stream
until the stream is terminated in that direction. Streams are an ordered
byte-stream abstraction, and they have no other structure within them. STREAM
frame boundaries are not expected to be preserved in retransmissions from the
sender or during delivery to the application at the receiver.When new data is to be sent on a stream, a sender MUST set the encapsulating
STREAM frame’s offset field to the stream offset of the first byte of this new
data. The first byte of data that is sent on a stream has the stream offset 0.
The largest offset delivered on a stream MUST be less than 2^64. A receiver
MUST ensure that received stream data is delivered to the application as an
ordered byte-stream. Data received out of order MUST be buffered for later
delivery, as long as it is not in violation of the receiver’s flow control
limits.The cryptographic handshake stream, Stream 1, MUST NOT be subject to congestion
control or connection-level flow control, but MUST be subject to stream-level
flow control. An endpoint MUST NOT send data on any other stream without
consulting the congestion controller and the flow controller.Flow control is described in detail in , and congestion control
is described in the companion document .Stream multiplexing has a significant effect on application performance if
resources allocated to streams are correctly prioritized. Experience with other
multiplexed protocols, such as HTTP/2 , shows that effective
prioritization strategies have a significant positive impact on performance.QUIC does not provide frames for exchanging priotization information. Instead
it relies on receiving priority information from the application that uses QUIC.
Protocols that use QUIC are able to define any prioritization scheme that suits
their application semantics. A protocol might define explicit messages for
signaling priority, such as those defined in HTTP/2; it could define rules that
allow an endpoint to determine priority based on context; or it could leave the
determination to the application.A QUIC implementation SHOULD provide ways in which an application can indicate
the relative priority of streams. When deciding which streams to dedicate
resources to, QUIC SHOULD use the information provided by the application.
Failure to account for priority of streams can result in suboptimal performance.Stream priority is most relevant when deciding which stream data will be
transmitted. Often, there will be limits on what can be transmitted as a result
of connection flow control or the current congestion controller state.Giving preference to the transmission of its own management frames ensures that
the protocol functions efficiently. That is, prioritizing frames other than
STREAM frames ensures that loss recovery, congestion control, and flow control
operate effectively.Stream 1 MUST be prioritized over other streams prior to the completion of the
cryptographic handshake. This includes the retransmission of the second flight
of client handshake messages, that is, the TLS Finished and any client
authentication messages.STREAM frames that are determined to be lost SHOULD be retransmitted before
sending new data, unless application priorities indicate otherwise.
Retransmitting lost STREAM frames can fill in gaps, which allows the peer to
consume already received data and free up flow control window.It is necessary to limit the amount of data that a sender may have outstanding
at any time, so as to prevent a fast sender from overwhelming a slow receiver,
or to prevent a malicious sender from consuming significant resources at a
receiver. This section describes QUIC’s flow-control mechanisms.QUIC employs a credit-based flow-control scheme similar to HTTP/2’s flow control
. A receiver advertises the number of octets it is prepared to
receive on a given stream and for the entire connection. This leads to two
levels of flow control in QUIC: (i) Connection flow control, which prevents
senders from exceeding a receiver’s buffer capacity for the connection, and (ii)
Stream flow control, which prevents a single stream from consuming the entire
receive buffer for a connection.A receiver sends WINDOW_UPDATE frames to the sender to advertise additional
credit by sending the absolute byte offset in the stream or in the connection
which it is willing to receive.The initial flow control credit is 65536 bytes for both the stream and
connection flow controllers.A receiver MAY advertise a larger offset at any point in the connection by
sending a WINDOW_UPDATE frame. A receiver MUST NOT renege on an advertisement;
that is, once a receiver advertises an offset via a WINDOW_UPDATE frame, it MUST
NOT subsequently advertise a smaller offset. A sender may receive WINDOW_UPDATE
frames out of order; a sender MUST therefore ignore any WINDOW_UPDATE that
does not move the window forward.A receiver MUST close the connection with a
QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA error () if the
peer violates the advertised stream or connection flow control windows.A sender MUST send BLOCKED frames to indicate it has data to write but is
blocked by lack of connection or stream flow control credit. BLOCKED frames are
expected to be sent infrequently in common cases, but they are considered useful
for debugging and monitoring purposes.A receiver advertises credit for a stream by sending a WINDOW_UPDATE frame with
the StreamID set appropriately. A receiver may use the current offset of data
consumed to determine the flow control offset to be advertised.
A receiver MAY send copies of a WINDOW_UPDATE frame in multiple packets in order
to make sure that the sender receives it before running out of flow control
credit, even if one of the packets is lost.Connection flow control is a limit to the total bytes of stream data sent in
STREAM frames on all streams contributing to connection flow control. A
receiver advertises credit for a connection by sending a WINDOW_UPDATE frame
with the StreamID set to zero (0x00). A receiver maintains a cumulative sum of
bytes received on all streams contributing to connection-level flow control, to
check for flow control violations. A receiver may maintain a cumulative sum of
bytes consumed on all contributing streams to determine the connection-level
flow control offset to be advertised.There are some edge cases which must be considered when dealing with stream and
connection level flow control. Given enough time, both endpoints must agree on
flow control state. If one end believes it can send more than the other end is
willing to receive, the connection will be torn down when too much data arrives.
Conversely if a sender believes it is blocked, while endpoint B expects more
data can be received, then the connection can be in a deadlock, with the sender
waiting for a WINDOW_UPDATE which will never come.On receipt of a RST_STREAM frame, an endpoint will tear down state for the
matching stream and ignore further data arriving on that stream. This could
result in the endpoints getting out of sync, since the RST_STREAM frame may have
arrived out of order and there may be further bytes in flight. The data sender
would have counted the data against its connection level flow control budget,
but a receiver that has not received these bytes would not know to include them
as well. The receiver must learn the number of bytes that were sent on the
stream to make the same adjustment in its connection flow controller.To avoid this de-synchronization, a RST_STREAM sender MUST include the final
byte offset sent on the stream in the RST_STREAM frame. On receiving a
RST_STREAM frame, a receiver definitively knows how many bytes were sent on that
stream before the RST_STREAM frame, and the receiver MUST use the final offset
to account for all bytes sent on the stream in its connection level flow
controller.Since streams are bidirectional, a sender of a RST_STREAM needs to know how many
bytes the peer has sent on the stream. If an endpoint receives a RST_STREAM
frame and has sent neither a FIN nor a RST_STREAM, it MUST send a RST_STREAM in
response, bearing the offset of the last byte sent on this stream as the final
offset.This document leaves when and how many bytes to advertise in a WINDOW_UPDATE to
the implementation, but offers a few considerations. WINDOW_UPDATE frames
constitute overhead, and therefore, sending a WINDOW_UPDATE with small offset
increments is undesirable. At the same time, sending WINDOW_UPDATES with large
offset increments requires the sender to commit to that amount of buffer.
Implementations must find the correct tradeoff between these sides to determine
how large an offset increment to send in a WINDOW_UPDATE.A receiver MAY use an autotuning mechanism to tune the size of the offset
increment to advertise based on a roundtrip time estimate and the rate at which
the receiving application consumes data, similar to common TCP implementations.If a sender does not receive a WINDOW_UPDATE frame when it has run out of flow
control credit, the sender will be blocked and MUST send a BLOCKED frame. A
BLOCKED frame is expected to be useful for debugging at the receiver. A
receiver SHOULD NOT wait for a BLOCKED frame before sending a
WINDOW_UPDATE, since doing so will cause at least one roundtrip of quiescence.
For smooth operation of the congestion controller, it is generally considered
best to not let the sender go into quiescence if avoidable. To avoid blocking a
sender, and to reasonably account for the possibiity of loss, a receiver should
send a WINDOW_UPDATE frame at least two roundtrips before it expects the sender
to get blocked.An endpoint that detects an error SHOULD signal the existence of that error to
its peer. Errors can affect an entire connection (see ),
or a single stream (see ).The most appropriate error code () SHOULD be included in the
frame that signals the error. Where this specification identifies error
conditions, it also identifies the error code that is used.Public Reset is not suitable for any error that can be signaled with a
CONNECTION_CLOSE or RST_STREAM frame. Public Reset MUST NOT be sent by an
endpoint that has the state necessary to send a frame on the connection.Errors that result in the connection being unusable, such as an obvious
violation of protocol semantics or corruption of state that affects an entire
connection, MUST be signaled using a CONNECTION_CLOSE frame
(). An endpoint MAY close the connection in this
manner, even if the error only affects a single stream.A CONNECTION_CLOSE frame could be sent in a packet that is lost. An endpoint
SHOULD be prepared to retransmit a packet containing a CONNECTION_CLOSE frame if
it receives more packets on a terminated connection. Limiting the number of
retransmissions and the time over which this final packet is sent limits the
effort expended on terminated connections.An endpoint that chooses not to retransmit packets containing CONNECTION_CLOSE
risks a peer missing the first such packet. The only mechanism available to an
endpoint that continues to receive data for a terminated connection is to send a
Public Reset packet.If the error affects a single stream, but otherwise leaves the connection in a
recoverable state, the endpoint can sent a RST_STREAM frame
() with an appropriate error code to terminate just the
affected stream.Stream 1 is critical to the functioning of the entire connection. If stream 1
is closed with either a RST_STREAM or STREAM frame bearing the FIN flag, an
endpoint MUST generate a connection error of type QUIC_CLOSED_CRITICAL_STREAM.Some application protocols make other streams critical to that protocol. An
application protocol does not need to inform the transport that a stream is
critical; it can instead generate appropriate errors in response to being
notified that the critical stream is closed.An endpoint MAY send a RST_STREAM frame in the same packet as a CONNECTION_CLOSE
frame.Error codes are 32 bits long, with the first two bits indicating the source of
the error code:
Application-specific error codes. Defined by each application-layer protocol.
Reserved for host-local error codes. These codes MUST NOT be sent to a peer,
but MAY be used in API return codes and logs.
QUIC transport error codes, including packet protection errors. Applicable to
all uses of QUIC.
Cryptographic error codes. Defined by the cryptographic handshake protocol
in use.This section lists the defined QUIC transport error codes that may be used in a
CONNECTION_CLOSE or RST_STREAM frame. Error codes share a common code space.
Some error codes apply only to either streams or the entire connection and have
no defined semantics in the other context.
Connection has reached an invalid state.
There were data frames after the a fin or reset.
Control frame is malformed.
Frame data is malformed.
Multiple final offset values were received on the same stream
The stream was cancelled
A stream that is critical to the protocol was closed.
The packet contained no payload.
STREAM frame data is malformed.
Received STREAM frame data is not encrypted.
Received a frame which is likely the result of memory corruption.
RST_STREAM frame data is malformed.
CONNECTION_CLOSE frame data is malformed.
GOAWAY frame data is malformed.
WINDOW_UPDATE frame data is malformed.
BLOCKED frame data is malformed.
PATH_CLOSE frame data is malformed.
ACK frame data is malformed.
Version negotiation packet is malformed.
Public RST packet is malformed.
There was an error decrypting.
There was an error encrypting.
The packet exceeded kMaxPacketSize.
The peer is going away. May be a client or server.
A stream ID was invalid.
A priority was invalid.
Too many streams already open.
The peer created too many available streams.
Received public reset for this connection.
Invalid protocol version.
The Header ID for a stream was too far from the previous.
Negotiable parameter received during handshake had invalid value.
There was an error decompressing data.
The connection timed out due to no network activity.
The connection timed out waiting for the handshake to complete.
There was an error encountered migrating addresses.
There was an error encountered migrating port only.
We received a STREAM_FRAME with no data and no fin flag set.
The peer received too much data, violating flow control.
The peer sent too much data, violating flow control.
The peer received an invalid flow control window.
The connection has been IP pooled into an existing connection.
The connection has too many outstanding sent packets.
The connection has too many outstanding received packets.
The QUIC connection has been cancelled.
Disabled QUIC because of high packet loss rate.
Disabled QUIC because of too many PUBLIC_RESETs post handshake.
Disabled QUIC because of too many timeouts with streams open.
QUIC timed out after too many RTOs.
A packet was received with the wrong encryption level (i.e. it should
have been encrypted but was not.)
This connection involved a version negotiation which appears to have been
tampered with.
IP address changed causing connection close.
Client address validation failed.
Stream frames arrived too discontiguously so that stream sequencer buffer
maintains too many gaps.
Connection closed because server hit max number of sessions allowed.An attacker receives an STK from the server and then releases the IP address on
which it received the STK. The attacker may, in the future, spoof this same
address (which now presumably addresses a different endpoint), and initiate a
0-RTT connection with a server on the victim’s behalf. The attacker then spoofs
ACK frames to the server which cause the server to potentially drown the victim
in data.There are two possible mitigations to this attack. The simplest one is that a
server can unilaterally create a gap in packet-number space. In the non-attack
scenario, the client will send an ACK frame with the larger value for largest
acknowledged. In the attack scenario, the attacker could acknowledge a packet
in the gap. If the server sees an acknowledgment for a packet that was never
sent, the connection can be aborted.The second mitigation is that the server can require that acknowledgments for
sent packets match the encryption level of the sent packet. This mitigation is
useful if the connection has an ephemeral forward-secure key that is generated
and used for every new connection. If a packet sent is encrypted with a
forward-secure key, then any acknowledgments that are received for them MUST
also be forward-secure encrypted. Since the attacker will not have the forward
secure key, the attacker will not be able to generate forward-secure encrypted
packets with ACK frames.IANA [SHALL add/has added] a registry for “QUIC Transport Parameters” under a
“QUIC Protocol” heading.The “QUIC Transport Parameters” registry governs a 16-bit space. This space is
split into two spaces that are governed by different policies. Values with the
first byte in the range 0x00 to 0xfe (in hexadecimal) are assigned via the
Specification Required policy . Values with the first byte 0xff are
reserved for Private Use .Registrations MUST include the following fields:
The numeric value of the assignment (registrations will be between 0x0000 and
0xfeff).
A short mnemonic for the parameter.
A reference to a publicly available specification for the value.The nominated expert(s) verify that a specification exists and is readily
accessible. The expert(s) are encouraged to be biased towards approving
registrations unless they are abusive, frivolous, or actively harmful (not
merely aesthetically displeasing, or architecturally dubious).The initial contents of this registry are shown in
.ValueParameter NameSpecification0x0000stream_fc_offset0x0001connection_fc_offset0x0002concurrent_streams0x0003idle_timeout0x0004truncate_connection_idQUIC Loss Detection and Congestion ControlGoogleGoogleUsing Transport Layer Security (TLS) to Secure QUICMozillasn3rdKey words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.The Transport Layer Security (TLS) Protocol Version 1.3This document specifies version 1.3 of the Transport Layer Security (TLS) protocol. TLS allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery.Packetization Layer Path MTU DiscoveryThis document describes a robust method for Path MTU Discovery (PMTUD) that relies on TCP or some other Packetization Layer to probe an Internet path with progressively larger packets. This method is described as an extension to RFC 1191 and RFC 1981, which specify ICMP-based Path MTU Discovery for IP versions 4 and 6, respectively. [STANDARDS-TRACK]Path MTU discoveryThis memo describes a technique for dynamically discovering the maximum transmission unit (MTU) of an arbitrary internet path. It specifies a small change to the way routers generate one type of ICMP message. For a path that passes through a router that has not been so changed, this technique might not discover the correct Path MTU, but it will always choose a Path MTU as accurate as, and in many cases more accurate than, the Path MTU that would be chosen by current practice. [STANDARDS-TRACK]Path MTU Discovery for IP version 6This document describes Path MTU Discovery for IP version 6. It is largely derived from RFC 1191, which describes Path MTU Discovery for IP version 4. [STANDARDS-TRACK]Guidelines for Writing an IANA Considerations Section in RFCsMany protocols make use of identifiers consisting of constants and other well-known values. Even after a protocol has been defined and deployment has begun, new values may need to be assigned (e.g., for a new option type in DHCP, or a new encryption or authentication transform for IPsec). To ensure that such quantities have consistent values and interpretations across all implementations, their assignment must be administered by a central authority. For IETF protocols, that role is provided by the Internet Assigned Numbers Authority (IANA).In order for IANA to manage a given namespace prudently, it needs guidelines describing the conditions under which new values can be assigned or when modifications to existing values can be made. If IANA is expected to play a role in the management of a namespace, IANA must be given clear and concise instructions describing that role. This document discusses issues that should be considered in formulating a policy for assigning values to a namespace and provides guidelines for authors on the specific text that must be included in documents that place demands on IANA.This document obsoletes RFC 2434. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Structured Streams: A New Transport AbstractionQUIC: Multiplexed Transport Over UDPGuide for Internet Standards WritersThis document is a guide for Internet standard writers. It defines those characteristics that make standards coherent, unambiguous, and easy to interpret. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.TCP Extensions for Multipath Operation with Multiple AddressesTCP/IP communication is currently restricted to a single path per connection, yet multiple paths often exist between peers. The simultaneous use of these multiple paths for a TCP/IP session would improve resource usage within the network and, thus, improve user experience through higher throughput and improved resilience to network failure.Multipath TCP provides the ability to simultaneously use multiple paths between peers. This document presents a set of extensions to traditional TCP to support multipath operation. The protocol offers the same type of service to applications as TCP (i.e., reliable bytestream), and it provides the components necessary to establish and use multiple TCP flows across potentially disjoint paths. This document defines an Experimental Protocol for the Internet community.Randomness Requirements for SecuritySecurity systems are built on strong cryptographic algorithms that foil pattern analysis attempts. However, the security of these systems is dependent on generating secret quantities for passwords, cryptographic keys, and similar quantities. The use of pseudo-random processes to generate secret quantities can result in pseudo-security. A sophisticated attacker may find it easier to reproduce the environment that produced the secret quantities and to search the resulting small set of possibilities than to locate the quantities in the whole of the potential number space.Choosing random quantities to foil a resourceful and motivated adversary is surprisingly difficult. This document points out many pitfalls in using poor entropy sources or traditional pseudo-random number generation techniques for generating such quantities. It recommends the use of truly random hardware techniques and shows that the existing hardware on many systems can be used for this purpose. It provides suggestions to ameliorate the problem when a hardware solution is not available, and it gives examples of how large such quantities need to be for some applications. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Transport Layer Security (TLS) Application-Layer Protocol Negotiation ExtensionThis document describes a Transport Layer Security (TLS) extension for application-layer protocol negotiation within the TLS handshake. For instances in which multiple application protocols are supported on the same TCP or UDP port, this extension allows the application layer to negotiate which protocol will be used within the TLS connection.Hypertext Transfer Protocol Version 2 (HTTP/2)This specification describes an optimized expression of the semantics of the Hypertext Transfer Protocol (HTTP), referred to as HTTP version 2 (HTTP/2). HTTP/2 enables a more efficient use of network resources and a reduced perception of latency by introducing header field compression and allowing multiple concurrent exchanges on the same connection. It also introduces unsolicited push of representations from servers to clients.This specification is an alternative to, but does not obsolete, the HTTP/1.1 message syntax. HTTP's existing semantics remain unchanged.The original authors of this specification were Ryan Hamilton, Jana Iyengar, Ian
Swett, and Alyssa Wilk.The original design and rationale behind this protocol draw significantly from
work by Jim Roskind . In alphabetical order, the contributors to
the pre-IETF QUIC project at Google are: Britt Cyr, Jeremy Dorfman, Ryan
Hamilton, Jana Iyengar, Fedor Kouranov, Charles Krasic, Jo Kulik, Adam Langley,
Jim Roskind, Robbie Shade, Satyam Shekhar, Cherie Shi, Ian Swett, Raman Tenneti,
Victor Vasiliev, Antonio Vicente, Patrik Westin, Alyssa Wilk, Dale Worley, Fan
Yang, Dan Zhang, Daniel Ziegler.Special thanks are due to the following for helping shape pre-IETF QUIC and its
deployment: Chris Bentzel, Misha Efimov, Roberto Peon, Alistair Riddoch,
Siddharth Vijayakrishnan, and Assar Westerlund.This document has benefited immensely from various private discussions and
public ones on the quic@ietf.org and proto-quic@chromium.org mailing lists. Our
thanks to all.RFC Editor’s Note: Please remove this section prior to publication of a
final version of this document.Issue and pull request numbers are listed with a leading octothorp.Defined short and long packet headers (#40, #148, #361)Defined a versioning scheme and stable fields (#51, #361)Define reserved version values for “greasing” negotiation (#112, #278)The initial packet number is randomized (#35, #283)Narrow the packet number encoding range requirement (#67, #286, #299, #323,
#356)Defined client address validation (#52, #118, #120, #275)Define transport parameters as a TLS extension (#122)SCUP and COPT parameters are no longer valid (#116, #117)Transport parameters for 0-RTT are either remembered from before, or assume
default values (#126)The server chooses connection IDs in its final flight (#119, #349, #361)The server echoes the Connection ID and packet number fields when sending a
Version Negotiation packet (#133, #295, #244)Definied a minimum packet size for the initial handshake packet from the
client (#69, #136, #139, #164)Path MTU Discovery (#64, #106)The initial handshake packet from the client needs to fit in a single packet
(#338)Forbid acknowledgment of packets containing only ACK and PADDING (#291)Require that frames are processed when packets are acknowledged (#381, #341)Removed the STOP_WAITING frame (#66)Don’t require retransmission of old timestamps for lost ACK frames (#308)Clarified that frames are not retransmitted, but the information in them can
be (#157, #298)Error handling definitions (#335)Split error codes into four sections (#74)Forbid the use of Public Reset where CONNECTION_CLOSE is possible (#289)Define packet protection rules (#336)Require that stream be entirely delivered or reset, including acknowledgment
of all STREAM frames or the RST_STREAM, before it closes (#381)Remove stream reservation from state machine (#174, #280)Only stream 0 does not contributing to connection-level flow control (#204)Stream 1 counts towards the maximum concurrent stream limit (#201, #282)Remove connection-level flow control exclusion for some streams (except 1)
(#246)RST_STREAM affects connection-level flow control (#162, #163)Flow control accounting uses the maximum data offset on each stream, rather
than bytes received (#378)Moved length-determining fields to the start of STREAM and ACK (#168, #277)Added the ability to pad between frames (#158, #276)Remove error code and reason phrase from GOAWAY (#352, #355)GOAWAY includes a final stream number for both directions (#347)Error codes for RST_STREAM and CONNECTION_CLOSE are now at a consistent offset
(#249)Defined priority as the responsibility of the application protocol (#104,
#303)Replaced DIVERSIFICATION_NONCE flag with KEY_PHASE flagDefined versioningReworked description of packet and frame layoutError code space is divided into regions for each componentUse big endian for all numeric valuesAdopted as base for draft-ietf-quic-tls.Updated authors/editors list.Added IANA Considerations section.Moved Contributors and Acknowledgments to appendices.